[Blink] Bkink pro crash

Adrian Georgescu ag at ag-projects.com
Mon Oct 20 22:47:30 CEST 2014


Can you send the crash report?

Adrian

On 20 Oct 2014, at 18:45, Daniel Guevara <dguevara at clearcom.mx> wrote:

> Hello , 
> 
> I am experimenting Blink crashes since latest version 4.0.1 in MACOS 10.9.4 .
> To reproduced the problem this is the scenario : 
> i Have an account to an internal Asterisk V1.8 , when i receive a call from an another internal extension the blink closed without reason.
> I have no problem in outgoing calls to the asterisk only inbound.
> This is the trace i had in the sip debug asterisk.
> 
> nxtphone*CLI> 
>   == Using SIP RTP CoS mark 5
>     -- Executing [1030 at anexos:1] Gosub("SIP/1004-00000261", "std-exten,~~s~~,1(1030,"SIP")") in new stack
>     -- Executing [~~s~~@std-exten:1] MSet("SIP/1004-00000261", "LOCAL(ext)=1030") in new stack
>     -- Executing [~~s~~@std-exten:2] MSet("SIP/1004-00000261", "LOCAL(dev)="SIP"") in new stack
>     -- Executing [~~s~~@std-exten:3] MSet("SIP/1004-00000261", "LOCAL(~~EXTEN~~)=~~s~~") in new stack
>     -- Executing [~~s~~@std-exten:4] MSet("SIP/1004-00000261", "LOCAL(~~EXTEN~~)=~~s~~") in new stack
>     -- Executing [~~s~~@std-exten:5] Set("SIP/1004-00000261", "CHANNEL(language)=es") in new stack
>     -- Executing [~~s~~@std-exten:6] MSet("SIP/1004-00000261", "DYNAMIC_FEATURES=automon") in new stack
>     -- Executing [~~s~~@std-exten:7] Dial("SIP/1004-00000261", "SIP/1030,20,TrtWw") in new stack
>   == Using SIP RTP CoS mark 5
> Audio is at 18978
> Adding codec 0x100 (g729) to SDP
> Adding codec 0x8 (alaw) to SDP
> Adding codec 0x2 (gsm) to SDP
> Adding codec 0x4 (ulaw) to SDP
> Adding codec 0x1000 (g722) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> Reliably Transmitting (no NAT) to 192.168.1.188:55605:
> INVITE sip:86302491 at 192.168.1.188:55605 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK4e269302
> Max-Forwards: 70
> From: "1004" <sip:1004 at 192.168.1.3>;tag=as41241883
> To: <sip:86302491 at 192.168.1.188:55605>
> Contact: <sip:1004 at 192.168.1.3:5060>
> Call-ID: 1c48ed916eefbe597874385a268ec938 at 192.168.1.3:5060
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 1.8.15-cert1
> Date: Mon, 20 Oct 2014 19:49:33 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces, timer
> Content-Type: application/sdp
> Content-Length: 382
> 
> v=0
> o=root 225996422 225996422 IN IP4 192.168.1.3
> s=Asterisk PBX 1.8.15-cert1
> c=IN IP4 192.168.1.3
> t=0 0
> m=audio 18978 RTP/AVP 18 8 3 0 9 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> 
> ---
>     -- Called SIP/1030
> Retransmitting #1 (no NAT) to 192.168.1.188:55605:
> INVITE sip:86302491 at 192.168.1.188:55605 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK4e269302
> Max-Forwards: 70
> From: "1004" <sip:1004 at 192.168.1.3>;tag=as41241883
> To: <sip:86302491 at 192.168.1.188:55605>
> Contact: <sip:1004 at 192.168.1.3:5060>
> Call-ID: 1c48ed916eefbe597874385a268ec938 at 192.168.1.3:5060
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 1.8.15-cert1
> Date: Mon, 20 Oct 2014 19:49:33 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces, timer
> Content-Type: application/sdp
> Content-Length: 382
> 
> v=0
> o=root 225996422 225996422 IN IP4 192.168.1.3
> s=Asterisk PBX 1.8.15-cert1
> c=IN IP4 192.168.1.3
> t=0 0
> m=audio 18978 RTP/AVP 18 8 3 0 9 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> 
> ---
> 
> <--- SIP read from UDP:192.168.1.188:55605 --->
> SIP/2.0 500 Internal Server Error
> Via: SIP/2.0/UDP 192.168.1.3:5060;received=192.168.1.3;branch=z9hG4bK4e269302
> Call-ID: 1c48ed916eefbe597874385a268ec938 at 192.168.1.3:5060
> From: "1004" <sip:1004 at 192.168.1.3>;tag=as41241883
> To: <sip:86302491 at 192.168.1.188>;tag=z9hG4bK4e269302
> CSeq: 102 INVITE
> Server: Blink Pro 4.0.1 (MacOSX)
> Content-Length: 0
> 
> <------------->
> --- (8 headers 0 lines) ---
>     -- Got SIP response 500 "Internal Server Error" back from 192.168.1.188:55605
> Transmitting (no NAT) to 192.168.1.188:55605:
> ACK sip:86302491 at 192.168.1.188:55605 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK4e269302
> Max-Forwards: 70
> From: "1004" <sip:1004 at 192.168.1.3>;tag=as41241883
> To: <sip:86302491 at 192.168.1.188:55605>;tag=z9hG4bK4e269302
> Contact: <sip:1004 at 192.168.1.3:5060>
> Call-ID: 1c48ed916eefbe597874385a268ec938 at 192.168.1.3:5060
> CSeq: 102 ACK
> User-Agent: Asterisk PBX 1.8.15-cert1
> Content-Length: 0
> 
> 
> ---
>     -- SIP/1030-00000262 is circuit-busy
>   == Everyone is busy/congested at this time (1:0/1/0)
>     -- Executing [~~s~~@std-exten:8] Goto("SIP/1004-00000261", "sw_36_CONGESTION,10") in new stack
>     -- Goto (std-exten,sw_36_CONGESTION,10)
>     -- Executing [sw_36_CONGESTION at std-exten:10] VoiceMail("SIP/1004-00000261", "1030 at default,ug(11)") in new stack
> Really destroying SIP dialog '1c48ed916eefbe597874385a268ec938 at 192.168.1.3:5060' Method: INVITE
>     -- <SIP/1004-00000261> Playing 'vm-theperson.ulaw' (language 'es')
>     -- <SIP/1004-00000261> Playing 'digits/1.ulaw' (language 'es')
>     -- <SIP/1004-00000261> Playing 'digits/0.ulaw' (language 'es')
>     -- <SIP/1004-00000261> Playing 'digits/3.ulaw' (language 'es')
>     -- <SIP/1004-00000261> Playing 'digits/0.ulaw' (language 'es')
>   == Spawn extension (std-exten, sw_36_CONGESTION, 10) exited non-zero on 'SIP/1004-00000261'
>     -- Executing [h at std-exten:1] Goto("SIP/1004-00000261", "9991") in new stack
>     -- Goto (std-exten,h,9991)
>     -- Executing [h at std-exten:9991] Set("SIP/1004-00000261", "~~parentcxt~~=anexos") in new stack
>     -- Executing [h at std-exten:9992] GotoIf("SIP/1004-00000261", "0?9996") in new stack
>     -- Executing [h at std-exten:9993] GotoIf("SIP/1004-00000261", "0?9994:9996") in new stack
>     -- Goto (std-exten,h,9996)
>     -- Executing [h at std-exten:9996] NoOp("SIP/1004-00000261", "") in new stack
> 
> Daniel Guevara
> _______________________________________________
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> Blink at lists.ag-projects.com
> http://lists.ag-projects.com/mailman/listinfo/blink

--
Adrian



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